I've recently signed to use voip.co.uk as a bit of an experiment. I'm
using Ubuntu Linux and the Linphone package. I can call PSTN numbers but I
get cut off after about 100 seconds or so. Below is my configuration for
Linphone. Does this look right? It seems to work, but I had to play around
to get it to do so. Oh, I've replaced my id with nnnnnn and my password with ******

Regards,
Tony.

[net]
con_type=3
use_nat=0
nat_address=81.86.172.58

[sip]
sip_port=5060
guess_hostname=0

use_info=0
use_ipv6=0
default_proxy=0

[rtp]
audio_rtp_port=7078
video_rtp_port=9078
audio_jitt_comp=60
video_jitt_comp=60

[sound]
dev_id=1
rec_lev=80
play_lev=80
source=m
local_ring=/usr/share/sounds/linphone/rings/oldphone.wav
remote_ring=/usr/share/sounds/linphone/ringback.wav

[video]
enabled=0
show_local=1

[audio_codec_0]
mime=PCMU
rate=8000
enabled=1

[audio_codec_1]
mime=GSM
rate=8000
enabled=1

[audio_codec_2]
mime=PCMA
rate=8000
enabled=1

[audio_codec_3]
mime=speex
rate=8000
enabled=1

[audio_codec_4]
mime=speex
rate=16000
enabled=1

[audio_codec_5]
mime=1015
rate=8000
enabled=1

[proxy_0]
reg_proxy=siproxy.voip.co.uk

reg_expires=120
reg_sendregister=1
publish=1

[auth_info_0]
username=nnnnn
passwd=*******
realm="voip.co.uk"

[auth_info_1]
username=nnnnnn
passwd=******
realm="80.249.108.5"

On Mon, 16 Jan 2006 22:51:05 +0000, Tony Arnold wrote:

I've recently signed to use voip.co.uk as a bit of an experiment. I'm
using Ubuntu Linux and the Linphone package. I can call PSTN numbers but I
get cut off after about 100 seconds or so. Below is my configuration for
Linphone. Does this look right? It seems to work, but I had to play around
to get it to do so. Oh, I've replaced my id with nnnnnn and my password with ******

Regards,
Tony.

[net]
con_type=3
use_nat=0
nat_address=81.86.172.58

[sip]
sip_port=5060
guess_hostname=0

use_info=0
use_ipv6=0
default_proxy=0

[rtp]
audio_rtp_port=7078
video_rtp_port=9078
audio_jitt_comp=60
video_jitt_comp=60

[sound]
dev_id=1
rec_lev=80
play_lev=80
source=m
local_ring=/usr/share/sounds/linphone/rings/oldphone.wav
remote_ring=/usr/share/sounds/linphone/ringback.wav

[video]
enabled=0
show_local=1

[audio_codec_0]
mime=PCMU
rate=8000
enabled=1

[audio_codec_1]
mime=GSM
rate=8000
enabled=1

[audio_codec_2]
mime=PCMA
rate=8000
enabled=1

[audio_codec_3]
mime=speex
rate=8000
enabled=1

[audio_codec_4]
mime=speex
rate=16000
enabled=1

[audio_codec_5]
mime=1015
rate=8000
enabled=1

[proxy_0]
reg_proxy=siproxy.voip.co.uk

reg_expires=120
reg_sendregister=1
publish=1

[auth_info_0]
username=nnnnn
passwd=*******
realm="voip.co.uk"

[auth_info_1]
username=nnnnnn
passwd=******
realm="80.249.108.5"



Your lucky! I'm using a Sipura SPA-3000 and I get disconnected after about
45-50 seconds! Have reported it to voip.co.uk.

Tony Arnold wrote:

I've recently signed to use voip.co.uk as a bit of an experiment. I'm
using Ubuntu Linux and the Linphone package.
snip

Whatever you do with Linphone, make sure you use it with ALSA instead of
OSS, as I've found it to be a complete and utter pig with OSS.

Might be worth watching your network traffic with gkrellm or iptraf to see
which side of the call gets cut off. Also, drop into an Asterisk console
with 'asterisk -rc' and watch for error messages. Whilst in the console,
'sip show peers', 'sip debug' and 'sip no debug' can be helpful.
And one more thing, I found creating an exten that lets one record a
message and play it back, helpful for isolating issues with particual SIP
peers.

alexd
--
http://ale.cx/ (AIM:troffasky) )
19:23:22 up 1 day, 23:38, 2 users, load average: 0.00, 0.05, 0.07
This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK

Thanks. I've had a response from the support people at voip.co.uk saying
they are aware of this problem with some devices. They are planning a
software upgrade to address the problem this week-end, so hopefully on
Monday all will be well again!

Regards,
Tony.

On Tue, 17 Jan 2006 19:33:04 +0000, alexd
wrote:

Tony Arnold wrote:

I've recently signed to use voip.co.uk as a bit of an experiment. I'm
using Ubuntu Linux and the Linphone package.
snip

Whatever you do with Linphone, make sure you use it with ALSA instead of
OSS, as I've found it to be a complete and utter pig with OSS.

Might be worth watching your network traffic with gkrellm or iptraf to see
which side of the call gets cut off. Also, drop into an Asterisk console
with 'asterisk -rc' and watch for error messages. Whilst in the console,
'sip show peers', 'sip debug' and 'sip no debug' can be helpful.
And one more thing, I found creating an exten that lets one record a
message and play it back, helpful for isolating issues with particual SIP
peers.

alexd